Hi everybody,
I have a sever inside a DMZ with a private IP address (10.20.30.40)
reachable from the outside with a Public IP (90.80.70.60) Natted 1:1 to
the private IP.
I want to setup mediaproxy on this server, but since I cannot configure
it with the public IP, in the resulting SDP messages the Private IP will
appear .
Can I use the function:
sdp_mangle_ip("10.20.30.40","90.80.70.60");
to correct this behaviour?
Or there is another way to run mediaproxy in this scenario?
Thanks in advance,
Cosimo
FYI
-------- Original Message --------
Subject: [SR-Dev] git repository online
Date: Tue, 18 Nov 2008 02:18:34 +0100
From: Andrei Pelinescu-Onciul <andrei(a)iptel.org>
To: sr-dev(a)lists.sip-router.org
As you probably have noticed the git repository for sip-router is now
online.
It was started from ser core and tm.
git urls:
git://git.sip-router.org/sip-router
(read only)
http://git.sip-router.org/sip-router
(read only, slower, git://... recomended)
ssh://git.sip-router.org/sip-router
(read write but account on git.sip-router.org needed)
Web interface: http://git.sip-router.org/cgi-bin/gitweb.cgi
Branches acl: the repository is setup so that you can write/commit
only to the following branches:
<your_username>/.*
tmp/.*
master
We'll add access to more branches as needed (I've added this rules to
avoid creating lots of "root" branches by mistake, e.g. git pull
from a local branch without a refspec).
Please create the commit messages following the git convention
(start with one short line, preferably less then 50 chars summarizing
the changes, then one empty line and then a more detailed description).
See:
http://www.kernel.org/pub/software/scm/git/docs/user-manual.html#creating-g…http://www.tpope.net/node/106
Quick git introduction:
# clone the repo into the sip-router directory
git clone git://git.sip-router.org/sip-router
cd sip-router
git branch -a # show all branches
# create a local branch based on origin/andrei/mod_f_params
git checkout --track -b my_branch origin/andrei/mod_f_params
git log # see commit log
git log --pretty=oneline # only the first line from each commit
vim Makefile.defs
git add Makefile.defs
git commit # or git commmit -a to skip the add step and commit all
# changed files
# push changes back (note you need ssh access for this)
# note: the next command will fail, unless your username is andrei :-)
# publish changes on the local "my_branc" to the remote
# andrei/mod_f_params
git push origin my_branch:andrei/mod_f_params
# or push them into the master branch if you are sure about them
git push origin my_branch:master
For more info about git try:
http://git.or.cz/gitwiki/GitDocumentation
and if you want to know how it works:
http://eagain.net/articles/git-for-computer-scientists/
(highly recommended)
CVS access:
CVS_SERVER="git cvsserver" cvs \
-d :ext:andrei@git.sip-router.org/sip-router co master
(instead of master you can use any other branch name that doesn't
include a '/')
The initial checkout takes a while, but after that cvs update seems fast
enough.
SVN access: unfortunately we do not have it. It looks like the
git svnserver is in the project stage (there is some python code but not
runnable yet). There is a way around it, but it would be quite complex
to setup: a git to svn gateway and then svn to git imports.
Other repositories on git.sip-router.org:
ser - entire ser updated from cvs
ser_core - only ser_core, automatically updated from cvs, read-only
test - test repository, contains an early sip-router version, it has the
same access rules as sip-router (if you have something you want
to test, please use it rather then trying directly on
sip-router).
Special thanks go to Jan, who not only did setup git.sip-router.org
(including automatic cvs sync for some of the repos), but he's also
hosting it on one of his private machines.
Andrei
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--
Daniel-Constantin Mierla
http://www.asipto.com
I'm having an issue with 1.3.2 where it's adding "100rel" to the Supported
header when it wasn't previously there. I'm sure this is something I'm
doing wrong, but I can't tell what it would be.
Sorry ¡ i forgot to put the subject.
Hello list.
I'm having some problems trying to build a counter for my kamailio.cfg file.
This is what I have..
main()
...
load avp from radius}
var(i)=0;
route(3);
...
route[1] {
t_on_failure("1");
if (!t_relay()) {
sl_reply_error();
};
xlog("L_INFO","[$ci] $rm t_relay a $ru");
exit;
}
route(3)
...
$var(nr)=$avp(s:caller_nr);
If ( var(i) < $var(nr) );
$var(i) = $var(i) + 1;
route(1);
else
exit;
failure_route[1] {
xlog("L_INFO", "[$ci] $rm:$ru failure_route\n");
if (t_was_cancelled()) {
xlog("L_INFO", "[$ci] $rm:$ru t_was_cancelled en failure_route\n");
exit;
}
if (!next_gw()) {
xlog("L_INFO", "[$ci] No hay mas gateways para $rm:$ou\n");
route(3);
} else {
xlog("L_INFO", "XXXXX");
}
t_on_failure("1");
t_relay();
}
After I restart the service the first time a call arrives the counter seems to go OK, the counter goes from 0 to 2 (if the caller_nr=3). But the second time the $var(i) at the main and route(3) it's = 0, but in the failure_route it keeps the value = 3. What I'm doing wrong?
Can someone help me here?
Thanks
Ricardo Martinez.-
Hi all,
I installed OpenSer on my Centos machine and everything worked fine...I need
now to install rtpproxy on the same machine but did not find any good
documentation that can help me about that...
I need to know please from where I can download the rtpproxy package and how
I can configure it
Regards
Hi All
If I have
modparam("usrloc", "db_mode", 0)
I can make calls between two users that are currently using my openser
installation.
If I set
modparam("usrloc", "db_mode", 1)
or
modparam("usrloc", "db_mode", 2)
It does not work any more and I get 408 errors, although the entries are
written to the location table in the database and openserctl ul show
-brief does show the users.
Any ideas ?
Thanks
Hi list,
I currently don¹t proxy audio nor have any plans to for all calls, except
for the "302 Redirect", where I need kamailio to become a b2bua with
RTPproxy (or for the record CALEA).
However, I can¹t seem to get the ³if² statement down on where
³engage_media_proxy² gets called without doing it for all outbound calls.
What is the best way to identify a call that was from a SIP client that
issued a 302?
My last effort was:
if ( uri == mysqlf && is_method("INVITE") ){
But that seemed to get every call. Any guidance would be appreciated,
thanks.
-graham
well put dan! I really don't understand why you would bother bringing this
to our attention at all Henning.
On Fri, Nov 14, 2008 at 11:12 PM, Dan Pascu <dan(a)ag-projects.com> wrote:
>
> I fail to see what worries you if all is fine and dandy in paradise. Don't
> mind us, we are just a bunch of loonies with weird ideas anyway.
>
> On Friday 14 November 2008, Henning Westerholt wrote:
> > Hello all,
> >
> > recently some statement came to my attention that "there is a common
> > consent that the current design/architecture of [..] OpenSER (inherited
> > from SER) is no longer able to deliver and to meet the present
> > requirements and demands".
> >
> > I don't want to argue that much about this opinion, in fact the demands
> > to a Voice over IP solution depends very much on the certain setup. But
> > i want to share some details from my experiences in developing and
> > operating a big VoIP infrastructure here at 1&1.
> >
> > We've about 2 million customers on our platform, that uses over 5
> > million individual numbers and terminate about 1 billion minutes per
> > month. We're able to provide a good service with the actual
> > architecture of OpenSER without any real problems. Of course there is
> > always some room for improvements, but so far the main challenges we
> > faced were not in the scalability or performance areas. More important
> > issues are e.g. the inherent complexity of the SIP protocol and the
> > maintainance of a good quality assurance and integration process.
> >
> > We started some years ago with OpenSER 0.9.5, which we then extended a
> > lot in house. For example we implemented more than 25 own modules, a
> > own path implementation, did a lot of bug fixing and workarounds for
> > certain problems we've found. We're able to reduce this amount of
> > proprietary code a lot in the past, because of progress in the OpenSER
> > development, intregration of our "key" modules and a lot of other
> > contributions. We're using now something between OpenSER 1.3 and
> > Kamailio 1.4 with only a few private extensions.
> >
> > So in my opinion the actual design of our server is not "[..] an
> > inevitable dead-end that needs to be avoided.". I rather think that
> > we'll be able with continuing improvements to tackle the upcoming
> > challenges well, especially as we will work together in the future with
> > the SER developers in improving important areas of this software.
> >
> > But this is just my personal opinion, everybody is of course free to
> > have their own position.
> >
> > With best regards,
> >
> > Henning Westerholt
>
>
>
> --
> Dan
>
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>
